| Index: webrtc/modules/audio_processing/audio_processing_impl.h
|
| diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
|
| index 6f237ae4dc84df14917777fdefd6041106875d65..bbd17191585037e9fa739a1390f5013023a4ba8e 100644
|
| --- a/webrtc/modules/audio_processing/audio_processing_impl.h
|
| +++ b/webrtc/modules/audio_processing/audio_processing_impl.h
|
| @@ -134,6 +134,7 @@ class AudioProcessingImpl : public AudioProcessing {
|
| int StartDebugRecording(FILE* handle) override;
|
| int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
|
| int StopDebugRecording() override;
|
| + void UpdateHistogramsOnCallEnd() override;
|
| EchoCancellation* echo_cancellation() const override;
|
| EchoControlMobile* echo_control_mobile() const override;
|
| GainControl* gain_control() const override;
|
| @@ -210,6 +211,8 @@ class AudioProcessingImpl : public AudioProcessing {
|
| bool was_stream_delay_set_;
|
| int last_stream_delay_ms_;
|
| int last_aec_system_delay_ms_;
|
| + int stream_delay_jumps_;
|
| + int aec_system_delay_jumps_;
|
|
|
| bool output_will_be_muted_ GUARDED_BY(crit_);
|
|
|
|
|